With the recent developments in communication networks, an increasing demand for reducing costs of communication has arisen. One approach for reducing costs has lead to the development of the so called Voice over IP feature (VoIP) (also known as IP telephony) which is currently becoming more and more popular. In brief, VoIP means that data which were transmitted before via the communication network are modified for being transmitted via an intermediate Internet Protocol (IP) based network. Thus, the transmitting path of the data within the non-IP based communication networks which is deemed to be the most expensive one is shortened as much as possible to reduce costs.
The current main benefits of the IP telephony concept are: Transmission savings which are achieved in two forms:                firstly, leased time divisional multiplexing (TDM) lines in the non-IP communication networks may (at least partly) be replaced with much cheaper IP connections, and        secondly, the amount of traffic in the IP network may further be cut down with compression techniques.        
In addition, a speech quality enhancement is achieved by implementing so called Tandem Free Operation (TFO) capabilities in used gateways.
The general network concept will be briefly explained with reference to FIG. 1. In this connection it is to be noted that the subsequent explanations will be given mainly with reference to a GSM communication network, while the invention is not limited thereto. Rather, also a UMTS (Universal Mobile Telecommunication System) network currently developed by the 3rd generation partnership project 3GPP or any other mobile or non-mobile communication network may be used instead. A terminal denoted by UE1 (user equipment and/or mobile station) communicates via a network NW_1 (not based on IP). In the network there are provided (not shown and in the order of a data flow originating from the terminal) an access network part consisting of e.g. at least one base station BS communicating with the terminal, the base station being controlled by a base station controller BSC, which in turn is controlled by a switching network part comprising e.g. at least one exchange such as a mobile services switching center MSC. According to GSM, after the BS or after the BSC there is provided a transcoding rate adaptation unit referred to as TRAU unit, performing a transcoding and rate adaptation of the data for further transmission. According to existing standards, the TRAU unit outputs a 64 kBit/s PCM (Pulse Code Modulated) signal to the MSC for further transmission.
The network NW_1 in turn is provided with and/or connected to a gateway MGW_1 (Media Gateway). Such a gateway may be represented by a Gateway MSC, i.e. a GMSC or by a separate gateway GW connected to a MSC.
Generally, a gateway provides an interface and/or interworking functionality such that the networks connected to the gateway may cooperated with each other. Stated in other words, gateway MGW_1 provides an interworking function for NW_1 (e.g. GSM) and IP_NW (e.g. the Internet). Thus, the PCM data arriving at the gateway are required to be converted and/or modified to IP compatible data (IP packets), since the IP based network IP_NW relies on a principle of transmitting data in units of packets.
The IP packet data are then transmitted via the IP based network IP_NW, a further gateway MGW_2, a further network NW_2 to a destination, i.e. terminal UE_2. The data transmission principle from the IP_NW up to the terminal UE_2 is similar to the transmission outlined above with the exception that it is performed in reverse order.
It is to be noted that the networks NW_1 and NW_2 may be different networks or the same network. Also, if different, they may rely on the same standard such as GSM or rely on different standards. Also, if they are the same network, the gateways may in most cases be located separately from each other, while it is not excluded that under certain circumstances the gateways MGW_1 and MGW_2 could be a single identical gateway
FIG. 2 shows in rough outline and in a strongly simplified manner some components of a gateway MGW (MGW_1 or MGW_2). It should be noted that the illustrated components are by far not the only components/functionalities of a gateway. Rather, only those components which will become necessary for understanding the present invention are shown in outline. It is also to be noted that the shown components may be provided as a separate module to be used as a plug-in unit in combination with remaining gateway components (not shown). Thus, with reference to FIG. 2, a 64 kBit/s PCM data stream arriving at the gateway from the network NW_1 or NW_2 (e.g. based on GSM standard) is processed by a signal processing device DSP (such as a digital signal processor, arranged between an PCM interface PCM-I/F and a IP interface IP-I/F. Likewise, data are processed by said signal processing device DSP prior to being output from said gateway (via an interface PCM-I/F) to said network NW_1 or NW_2.
The DSP performs various tasks such as echo cancellation, DTMF tone detection, jitter buffering, TFO (tandem free operation) control, and speech coding/decoding, as well as controlling the DSP (functional units thereof) as such. Each task may be represented by a specific means/unit in the DSP and/or by a specific processing carried out by the DSP, mostly in parallel.
For example, jitter buffering is provided in order that the variable IP network delay be compensated by buffering a number of the IP frames.
For all tasks to be performed by the DSP, the DSP has to rely on an overall processing power. Of the above mentioned task, the coding/decoding operation consumes the most part of the available processing power. Thus, for each channel, i.e. communication connection between communication partners such as the terminals UE_1 and UE_2, a codec means (coding/decoding means) has to be provided for and/or implemented in order to perform coding/decoding of the data transmitted between the communication partners.
Consequently, considering the limited overall processing power of the DSP, there results a limitation in the channel capacity of the DSP. Stated in other words, the number of channels that can be handled by the digital signal processor is limited. The number of channels that can be handled, i.e. the number of codecs implemented in the signal processor is set such that even under conditions of a worst case scenario for each channel the processing capacity of the signal processor is sufficient to successfully handle the coding/decoding for every channel.
Each codec of a respective channel is adapted to handle different channel types and may be controlled to change its operating mode according to the channel type. Examples for such channel types are the adaptive multi rate AMR, enhanced full rate EFR, full rate FR, half rate HR, and G.711, Wideband AMR, AMR TFO, EFR TFO, FR TFO, HR TFO Wideband AMR TFO, etc. Each channel type differs in the transmission capacity and hence in the processing capacity required for performing a codec operation for such a channel. The processing capacity is highest for Wideband AMR/AMR and decreases for EFR, FR, HR in this order down to the lowest required processing capacity for G.711.
In order to handle numerous channels, a previous approach resided in providing several DSPs and to distribute the channels to be handled by them by means of a controlled switch serving the DSPs provided for. This required extra hardware for control purposes of such a centralized resource management. However, even in such a case, the channel capacity remained limited and a further addition of DSPs was rather disadvantageous in terms of correspondingly increasing costs.